As wireless systems continue to evolve, communications between a mobile switching center (MSC) and its bases stations are moving to an Internet Protocol (IP) based transport mechanism. (As used herein, the term wireless systems refers to e.g., CDMA (code division multiple access), GSM (Global System for Mobile Communications), the proposed UMTS (Universal Mobile Telecommunications System), etc.) Given the nature of wireless communications, e.g., real-time voice, any IP-based transport needs to utilize a protocol that accommodates real-time applications.
One such protocol is the Real Time Protocol (RTP) (e.g., see H. Schulzrinne, R. Frederick, V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” RFC 1889). RTP is attractive since it is an available Internet Engineering Task Force (IETF) protocol for handling real-time streams. RTP traffic is encapsulated in UDP (user datagram protocol), and IP packets.
However, while RTP is designed to support real time applications, RTP makes no assumption on the ability of the underlying network to provide timely delivery or quality-of-service (QoS) guarantees. As such, RTP performs best when the underlying network is not heavily loaded and the applications using RTP can adapt to the underlying network conditions to some extent.
In addition, the very size of the packet payload in wireless applications also presents a problem when using RTP. For example, packets that transport voice are, in general, rather small compared to packets that transport mere data. ITU-T G723.1 specifies generation of a 20 byte speech packet at 30 ms intervals (e.g., see ITU-T Recommendation G.723.1 “Dual Rate Speech Coder for Multimedia Communications Transmitting At 5.3 and 6.3 Kbps,” 1995). Consequently, packets used to transport voice are subjected to a large overhead. For example a 10-byte voice packet transmitted using a User Datagram Protocol/Internet Protocol (UDP/IP) encapsulation incurs an overhead of 28 bytes (20 byte IP header, plus up to 8 bytes of UDP header), or 280%. In addition, if each application session (also referred to herein as an audio stream, or packet flow), requires use of one UDP session, the resulting large number of packets may create heavy packet processing load for any intermediate routers.
Fortunately, to improve transport efficiency, some multiplexing schemes have been proposed within the framework of RTP (e.g., see J. Rosenberg, “An RTP Payload Format for User Multiplexing,” work in progress, draft-ietf-avt-aggregation-00.txt; and B. Subbiah, S. Sengodan, “User Multiplexing in RTP payload between IP Telephony Gateway,” work in progress, draft-ietf-avt=mux-rtp-00.txt, August, 1998). An illustrative portion of a protocol stack, 30, using an RTP-based multiplexing scheme is shown in FIG. 1. Traffic is first multiplexed via the RTP Mux layer. RTP traffic is then encapsulated in UDP and IP packets. (It should be noted that other layers (not shown) also exist above and below. For example, below the IP layer sits the media access control (MAC) layer, which is on top of the physical layer, as known in the art.) However, none of these approaches address QoS.